Method and system for customer selected direct dialed voice-over-internet protocol (VOIP)

ABSTRACT

A Voice-over-Internet protocol (VoIP) communications network system that enables direct-dialed (single-stage) access to the Internet Protocol (IP) network from the circuit-switched network. Specifically, the VoIP network system includes a VoIP service implemented on a communications system which, after a customer number has been registered for the service, automatically recognizes calls from the registered customer&#39;s telephone number and determines if the call can be routed as a VoIP call over the IP network. In embodiments of the present invention, the customer can register for the VoIP service by selecting both a provider (PIC) and a calling plan or by only selecting a provider. The system can be implemented to handle intra-state, inter-state and international voice-band calls (for example, regular telephone calls, facsimile transmissions and modem initiated calls) using standard circuit-switched telephone lines, cable, twisted pair, digital subscriber line (DSL) and wireless.

CROSS REFERENCE TO RELATED APPLICATIONS

This application is a continuation of U.S. patent application Ser. No.09/599,947 filed on Jun. 23, 2000 now U.S. Pat. No. 7,046,658, entitled“METHOD AND SYSTEM FOR CUSTOMER SELECTED DIRECT DIALEDVOICE-OVER-INTERNET PROTOCOL (VoIP)” the contents of which areincorporated herein by reference.

TECHNICAL FIELD

The present invention relates generally to Voice-over-Internet protocol(VoIP) systems, and more particularly to direct-dialed, that is,“single-stage” VoIP systems.

BACKGROUND OF THE INVENTION

Voice-over-Internet protocol (VoIP) telephony systems deal with thetransmission of voice-band calls over a packet data network, such as acorporate intranet or the Internet. Using current VoIP systems to makelong distance calls offers a number of interesting advantages whencompared to traditional long distance telephone service. Some of theseadvantages include, for example, the ability to place long distancecalls as two local voice band calls using an IP data link between them,one local call at the calling number end and one at the destinationnumber end; and being able to manage a voice and data network as asingle network. Likewise, additional advantages include moving, addingand changing Internet Protocol (IP) phones is easier and cheaper thanregular telephones; providing new and integrated services includingintegrated messaging, bandwidth on demand, voice e-mails, “voiceportals” that provide interactive voice response access to systems suchas the Internet; and simplified setting up, tearing down andtransferring of phone calls.

Unfortunately, current VoIP systems have some significant disadvantagesas well, including no single-stage dialing as on a circuit-switchedPublic Switched Telephone Network (PSTN); only two- or three-stagedialing capabilities; and no integrated billing system that can detectand track network use and associate the use with a number for billingpurposes. “Single-stage” dialing, which is also known as direct-dialing,permits a caller to dial the desired destination number, the telephonenetwork automatically recognizes the telephone number from which thecaller initiated the call, the telephone network automaticallydetermines which calling plan(s) the caller has been registered androutes the call based on the caller's calling plan(s).

“Two-stage” or “three-stage” dialing both require the caller to firstcall an IP network access number, which can be either a local or anational number, and either a toll-free or a toll number. Next, fortwo-stage dialing, the system at the access number either automaticallydetects and recognizes the caller's number or the caller's phone serviceis programmed to automatically send the caller's account number and PINwhen the caller dials the access number; the system then connects thecaller to the system and the caller enters the desired destinationnumber. Alternatively, for three-stage dialing, after the system at theaccess number answers, the caller is usually prompted to enter thecaller's account number and PIN and, then, the caller enters the desireddestination number.

An example of both a two-stage and a three-stage dialing, prepaid VoIPcalling plan is AT&T's Connect 'N Save® service. In this two-stagedialing service, the customer enrolls in the service, prepays a setamount using a credit/debit card or a check, and signs up for theExpress Login feature at the customer's registered telephone number. TheExpress Login feature only operates from the customer's telephone numberand automatically identifies the customer's account number and PIN whenthe customer calls the access number from the customer's registeredtelephone number. After connecting to the service, the customer dialsthe desired destination number and waits for the call to connect withthe destination number. The Connect 'N Save® service has both local andnational access numbers that can be used to access the service. Anexample of three-stage dialing occurs in the Connect 'N Save® service ifthe customer either did not sign up for the Express Login feature or iscalling an access number from a telephone number other than the one thatwas used to sign up for the Connect 'N Save® service. In this case, thecustomer first dials the access number; second, enters the customer'saccount number and PIN; and third, dials the desired destination number.

FIG. 1, depicts a block diagram implementation of current two-stage andthree-stage voice-over-Internet protocol (VoIP) services. In FIG. 1,customer (calling party) telephones 115 and 116 are connected to a localaccess provider network 118, which is in turn connected to acircuit-switched, Public Switched Telephone Network (PSTN) 145 and to anIP gateway 120, which is communicatively linked to an IP Network 110.The IP gateway 120 provides both local and toll free number VoIP accessto the IP Network 110, however, single telephones are shown accessingthe IP gateway 120 for each of the local and toll free VoIP accessnumbers in FIG. 2 for reasons of clarity. The IP Network 110 iscommunicatively linked with another IP gateway 125, which provides VoIPservice to U.S. locations, and the IP gateway 125 is communicativelylinked to a destination telephone 135 through a destination local accessprovider network (not shown). Similarly, the IP Network 110 iscommunicatively linked with another IP gateway 130, which provides VoIPservice to international locations, and the IP gateway 130 iscommunicatively linked to a destination telephone 136 through anotherdestination local access provider network (not shown). The IP Network110 is communicatively linked with another IP gateway 140, which iscommunicatively linked to a second circuit-switched PSTN 145. When acalling party places a call to a destination telephone number that isnot served by any of the VoIP service IP gateways 125 and 130, the callis routed through the IP gateway 140 to the second PSTN 145 forcompletion as a circuit-switched telephone call.

In FIG. 1, when a caller desires to place a VoIP call the caller firstcalls one of the access numbers at the IP gateway 120 from one of thetelephones 115 and 116. These access numbers can be either a local or anational numbers, and either a toll-free or a toll number. Next, fortwo-stage dialing, the system at the IP gateway 120 either automaticallydetects and recognizes the caller's number or the caller's phone servicecan be programmed to automatically send the caller's account number andPIN when the caller dials the access number, connects the caller to thesystem and the caller enters the desired destination number.Alternatively, for three-stage dialing, after the system at the IPgateway 120 answers, the caller is usually prompted to enter thecaller's account number and PIN or credit/calling/debit card number andPIN and, then, the caller enters the desired destination telephonenumber. Upon receiving the destination telephone number, the IP gateway120 determines an appropriate destination IP gateway, for example,destination IP gateway 125, converts the call into packets and thenroutes the packets to the destination IP gateway 125 through the IPNetwork 110. The IP network 110 is configured to receive the packets andthen route the packets to the destination IP gateway 125 through thedestination local access provider network (not shown). The destinationIP gateway 125 receives the packets, reassembles the packets, convertsthe packets back to a voice-band call and sends the reassembledvoice-band call to the destination telephone number 135 through theother destination local access provider network (not shown).

Unfortunately, the currently-available two-stage and three-stage VoIPservices require the entry of multiple phone, account and PIN numbers touse the service, which is inefficient and unpleasant for customers.Another disadvantage of some currently-available two- and three-stageVoIP services is that they require either the purchase of new prepaidcalling cards or the replenishment of minimum prepaid account balancesbecause the calling party's regular telephone service billing system isnot connected to and does not communicate with the VoIP service.

Accordingly, a single-stage VoIP system is desired that enables the userto directly dial a destination number from a telephone attached to thecircuit-switched PSTN, where the call is automatically routed as a VoIPcall over the IP network and billed to the calling party's regulartelephone-bill account. Likewise, a provisioning system is desired thatreceives customer orders for the VoIP service, provisions the networkand billing systems based on the orders and maintains the operationaland informational synchronization between the network and billingsystems.

SUMMARY OF THE INVENTION

The present invention is directed to a single-stage VoIP system thatenables the user to directly dial a destination number from a telephoneattached to the circuit-switched PSTN and have the call automaticallyrouted as a VoIP call over the IP network and billed to the callingparty's regular telephone-bill account.

In an embodiment of the present invention, a method for routingdirect-dialed voice-band calls over an IP network includes receiving adirect-dialed voice-band call from a calling party telephone number, thedirect-dialed voice-band call being associated with a destinationnumber. The method further includes automatically routing thedirect-dialed voice-band call to the destination number as a VoIPtelephone call if the calling party telephone number is registered for aVoIP service and if the destination number of the direct-dialedtelephone call is accessible by the VoIP service.

In an embodiment of the present invention, a method for automaticallyprovisioning and maintaining a network system for routing direct-dialedvoice-band calls from a calling party telephone number over an IPnetwork includes receiving a VoIP service registration for the callingparty telephone number, generating at least one order record for thecalling party telephone number's VoIP service and storing the at leastone order record for the calling party telephone number's VoIP service.The method further includes managing the billing interaction for abilled account between at least one calling party telephone number and abilled telephone number; synchronizing changes made to the stored atleast one order record for the calling party telephone number's VoIPservice, between the network system and a billing system, due to callingparty activations, disconnections and changes; and processing at leastone call detail record including at least a terminating accessidentification (ID).

In an embodiment of the present invention, an apparatus includes a firstvoice-band switch and a database coupled to the first voice-band switch.In the apparatus, the first voice-band switch is configured to receive adirect-dialed voice-band call from a calling party's telephone numberand to automatically designate the direct-dialed voice-band call as aVoIP call.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a block diagram implementation of known two-stage andthree-stage voice-over-Internet protocol (VoIP) services.

FIG. 2 is a block diagram of the direct-dialed VoIP system according toan embodiment to the present invention.

FIG. 3 is a detailed block diagram of the direct-dialed VoIP system ofFIG. 2 which illustrates a direct-dialed VoIP apparatus according to anembodiment to the present invention.

FIG. 4 is a detailed functional flow diagram of the operation of theentire direct-dialed VoIP network system of FIG. 3, in accordance withan embodiment to the present invention.

FIG. 5 is a detailed functional flow diagram of the operation detailedin FIG. 4 for the direct-dialed VoIP apparatus of the entiredirect-dialed VoIP network system of FIG. 3 according to an embodimentto the present invention.

FIG. 6 is a detailed block diagram of an offline provisioning systemwhich is necessary for the proper operation of the direct-dialed VoIPservice according to an embodiment to the present invention.

FIG. 7 is a detailed functional flow diagram of the operation of ascalable offline provisioning system in FIG. 6 according to anembodiment to the present invention.

DETAILED DESCRIPTION

A direct-dialed voice-over-Internet protocol (VoIP) service is providedto registered customers so that the VoIP service is transparent to thecustomers when it is in operation. In accordance with an embodiment ofthe present invention, a customer registers (signs-up) for the VoIPservice (calling plan) and, once a customer registers for the service,then all of the customer's voice-band calls can be automatically routedthrough the VoIP service just as with other standard circuit-switchedcalling plans. Note that the registration of customers for the VoIPservice is not required by embodiments of the present invention becausethe VoIP service could be offered as the standard service that customerswould have to opt out of, that is, choose not to use, by registering foran alternate calling plan.

FIG. 2 is a block diagram of the direct-dialed VoIP system according toan embodiment to the present invention. In FIG. 2, a customer telephone200 is connected to an originating local-access-provider network 205 andassigned a telephone number (not shown), which is stored at theoriginating local-access-provider network 205. More generally, theoriginating local-access-provider network 205 may provide localvoice-band telecommunications services for numerous telephone numbers,however, only a single telephone 200, which is associated with a singletelephone number, is illustrated in FIG. 2 for reasons of clarity. Analternate embodiment of the present invention is contemplated in whichmultiple telephones, each associated with a unique telephone number, areconnected to and registered for the VoIP service. In another embodimentof the present invention, multiple telephones can be associated with thesingle telephone number registered for the VoIP service, for example,the multiple telephones can be extension phones that are all proximallylocated so that they can be connected to a single telephone line, whichprovides service for the single telephone number. The originatinglocal-access-provider network 205 may be communicatively linked with acircuit-switched, PSTN 210 when a calling party places a call to one ofdestination telephones 135, 136. The destination telephones can belocated both within and outside the service area of the originatinglocal-access-provider network 205. For example, calls outside the localservice area can include an intra-state long distance call, anout-of-state long distance call or an international long distance call.

The PSTN 210 is capable of providing service for numerous differentcalling plans and millions of different customers. In accordance withembodiments of the present invention, the PSTN 210 generally includes along distance telephone provider network, a local telephone serviceprovider and a Private Branch eXchange (PBX). PSTN 210 iscommunicatively linked with an IP gateway 215, and the IP gateway 215 iscommunicatively linked with the IP network 110. The PSTN 210 also iscommunicatively linked with other US and foreign destinations that arenot served by IP gateways. The IP network 110 is communicatively linkedwith an IP gateway 125 for US locations served by this IP gateway, andthe IP network 110 is also is communicatively linked with an IP gateway130 for foreign locations served by this IP gateway. Note that whileFIG. 2 and the above detail only show and describe single components forthe sake of clarity, each component can be representative of multiplecomponents. For example, multiple telephones (for example, telephone200) each can be communicatively linked to the local-access-providernetwork 205. Likewise, multiple originating local-access-providernetworks 205 can each be communicatively linked to the PSTN 210 of FIG.2, or each of a plurality of originating local-access-provider networks205 can each be communicatively linked to a separate PSTN 210. Similarmultiple configurations of the IP gateways 215, 125 and 130, the IPnetwork 110 and destination telephones 135 and 136 are also contemplatedin embodiments of the present invention.

In FIG. 2, in accordance with an embodiment of the present invention, acaller places a direct-dialed long-distance call from the telephone 200,that is, for example, the caller dials 1-555-123-4567. The call isreceived by the originating local-access-provider network 205 and thenrouted to the PSTN 210, because the “1-” dialing prefix identifies thecall as a long-distance call. After receiving the direct-dialedlong-distance call, the PSTN 210 determines if the call (or a portionthereof) can be routed as a VoIP call over the IP network 110. The callcan be routed as a VoIP call over the IP network 110 if the caller'snumber is registered for the VoIP service and the destination number isserved by the IP network. If the call can be routed as a VoIP call, thePSTN 210 initiates a VoIP billing record to track the duration and costof the VoIP call and, then, the PSTN 210 routes the call to an IPgateway 215. Once the IP gateway 215 has received the call, the IPgateway 215 determines a specific destination IP gateway 125 or 130based on the destination number, for example, IP gateway 125; convertsthe voice-band call setup information to packets, and then routes thepackets to the destination IP gateway 125 through the IP network 110. IPnetwork 110 is configured to receive the packets and then route thepackets to the destination IP gateway 125. Destination IP gateway 125receives the packets, reassembles the packets, converts the packets backto a voice-band call and sends the reassembled voice-band call to thedestination number 135 through a destination local access networkprovider (not shown).

In an alternate embodiment of the present invention, an originatinglocal-access-provider network can be configured to communicate directlywith an IP gateway and perform the same functions as described above forthe PSTN 210. Similarly, in another embodiment, an originatinglocal-access-provider network 205 can be configured to communicatedirectly with an IP gateway 215 for calls within the service area of theoriginating local-access-provider network 205 (communication link 206).

In another embodiment of the present invention, an originatinglocal-access-provider network can be a corporate telephone network orPrivate Branch eXchange (PBX) which is configured to communicatedirectly with an IP gateway and an IP network, which is a local areanetwork (LAN).

FIG. 3 depicts a detailed block diagram of the direct-dialed VoIP systemof FIG. 2 which illustrates a direct-dialed VoIP apparatus according toan embodiment to the present invention. In FIG. 3, elements that arecommon with FIG. 1 and FIG. 2 maintain their prior numbering schemes. InFIG. 3, telephone 200 is communicatively linked to the originatinglocal-access-provider network 205, which in turn is communicativelylinked to PSTN 210. In FIG. 3, in accordance with an embodiment of thepresent invention, PSTN 210 is shown to include a voice-band telephoneswitch 305, such as, for example, an Electronic Switching System Number4 (4ESS) Originating Assist Switch (OAS), which provides thecircuit-switched communication link with the originatinglocal-access-provider network 205. The PSTN 210 also is communicativelylinked to a database 310. Database 310 may be, for example which isimplemented in FIG. 3 as a Universal Subscriber Data Structure (USDS),containing, but not limited to, customer telephone number information,such as, registered network service(s) for the telephone number, acountry code listing of available destination IP gateways, and routinginstructions. The OAS 305 receives an incoming call from telephone 200and uses information associated with the incoming call and USDS 310information to determine if the incoming call is to be routed as a VoIPor a circuit-switched call.

The OAS 305 also may be communicatively linked to a second telephoneswitch 307, such as, for example, a 4ESS Handoff Assist Switch (HAS),which is used to route the calls to be routed as VoIP calls, and a thirdtelephone switch 315, such as, for example, a 4ESS InternationalSwitching Center (ISC), which is used to route the calls to be routed ascircuit-switched calls. In accordance with an embodiment of the presentinvention, the HAS 307 is communicatively linked to the originating IPgateway 215, which is, in turn, linked to an IP command center database325, which contains a listing of the specific IP gateways or complex ofgateways that serve the destination number. The IP gateway 215 uses thespecific IP gateway information from the IP command center database 325to determine the final routing instruction for the call, converts thevoice-band call and setup information to packets and then routes thepackets to the IP network 110. In addition to the routing function, theHAS 307 may create a billing record that is used to track and record thelength of the VoIP call to billed to the caller's standardtelephone-bill account. The HAS 307 was used, in this embodiment of thepresent invention, to consolidate and reduce the number of IP gateways215 and the number of connections that are actually made to the IPgateway 215 since the IP gateway 215 can only handle a limited number ofconnections. For example, in the experimental design, the IP gateway 215could only receive a single T-1 (Trunk Level 1) connection. Inaccordance with an embodiment of the present invention, the system isconfigured with 6 IP gateways 215 and each IP gateway 215 can receivefour (4) separate T-1 lines.

In an alternate embodiment of the present invention, an HAS is not usedand an OAS is directly and communicatively linked to an IP gateway andthe OAS also is modified to perform the functions of the HAS describedabove. However, in this embodiment of the present invention, the OAS canonly be located a finite distance (for example, approximately 900 miles)from the IP gateway.

The ISC 315 is communicatively linked to other circuit switches, atdestinations that are not served by IP gateways, to route standardcircuit-switched calls.

The IP network 110 routes the packets to the appropriate destination IPgateway 125, 130 where the packets are received, reassembled and thenconverted back to a voice-band call. IP gateway 125 is communicativelylinked to an Electronic Switching System Number 5 (5ESS) 330 and IPgateway 125 routes the reassembled voice-band call to the 5ESS 330. The5ESS 330 routes the reassembled voice-band call to a destinationlocal-access-provider network 335, which, in turn, routes thereassembled voice-band call to the destination telephone number 135. IPgateway 130 is directly and communicatively linked to anotherdestination local-access-provider network 340 and IP gateway 130 routesthe reassembled voice-band call to the destination local-access-providernetwork 340. The other destination local-access-provider network routesthe reassembled voice-band call to the destination telephone number 136.

In accordance with an embodiment of the present invention, an emergencybackup network is associated with the direct-dialed VoIP service and isconfigured to operate when, the destination IP gateway 125 or 130 goesdown, that is, becomes unavailable, after the direct-dialed call hasbeen routed to the originating IP gateway 215. In this embodiment of thepresent invention, if, after the direct-dialed call has been routed tothe originating IP gateway 215, the destination IP gateway 125 or 130goes down, the service automatically routes the direct-dialed call toanother IP gateway in the IP network 110 and then uses the Connect 'NSave® service to complete the call. The other IP gateway can be locatedanywhere in the IP network 110. In an embodiment of the present,preference is first given to the other IP gateway that is located thenearest to the destination IP gateway 125 or 130. If the nearest IPgateway is also unavailable, then the call is routed through an IPgateway, which serves as the main IP gateway for the Connect 'N Save®service, to a 4ESS switch that, then, forwards the call to the ESS ISC315 for completion as a circuit-switched call.

In FIG. 3, when a called party at a destination telephone numbertransmits a voice-band response back to the calling party, the aboveprocess is reversed and the transmitted voice-band response is routedthrough the local-access-provider network 335 or 340 and then to theappropriate IP gateway 125 or 130 where the voice-band response isconverted to packets and routed over the IP network 110 to theoriginating IP gateway 215. When the IP gateway 215 receives all of thepackets from the voice-band response, the IP gateway 215 reassembles thepackets in the correct order; converts the packets back to a voice-bandresponse and, then, routes the reassembled voice-band response to thePSTN 210 for routing through the local-access-provider network 205 tothe calling party at telephone number 200.

FIG. 4 depicts a detailed functional flow diagram of the operation ofthe entire direct-dialed VoIP network system of FIG. 3, in accordancewith an embodiment to the present invention. In FIG. 4, in block 405 adirect-dialed voice-band call is received by the OAS 305 from thecalling party's telephone number, where the direct-dialed voice-bandcall is associated with a destination telephone number. In block 410, atest is performed to determine if the calling party's telephone numberwas registered for the VoIP service. In an embodiment of the presentinvention, before the test can be performed, the OAS 305 sends thecalling party's telephone number and the associated destinationtelephone number to the USDS 310. After receiving these numbers, theUSDS 310 does a lookup in a database using the calling party's telephonenumber to determine if the calling party's telephone number has beenregistered for the VoIP service. If the calling party's telephone numberwas not registered for the VoIP service, then, the USDS 310 sets aterminating address value to indicate whichever service the callingparty is registered, for example, “PSTN”, or to a system determineddefault value if the calling party was not registered for any networkservices. The USDS 310, then returns the terminating address to the OAS305 where the OAS 305 determines that the terminating address valueindicates the voice-band call is to be routed as a circuit-switchednetwork call and, then, in block 412, the OAS 305 routes thedirect-dialed voice-band call to the ESS ISC 315 to continue routing thedirect-dialed voice-band call as a circuit-switched call through thePSTN. In block 414, a standard circuit-switched billing record isinitiated. If the calling party's telephone number was registered forthe VoIP service, then, in block 415, a test is performed to determineif the destination number of the voice-band call is served by an IPgateway. If the destination number of the voice-band call is not servedby an IP gateway, then, in block 412, the voice-band call is routed as acircuit-switched network call and, in block 414, a standardcircuit-switched billing record is initiated.

If the calling party's telephone number was registered for the VoIPservice, then, the USDS 310 then checks the destination telephone numberagainst an allowable number to determine if the destination telephone isserved by an IP gateway. If the destination telephone number is notserved by an IP gateway, the USDS 310 sets the terminating address toindicate a non-VoIP service, for example, “PSTN”, and then returns tothe OAS 305 the terminating address. The OAS 305, then, determines thatthe terminating address value indicates the voice-band call is to berouted as a circuit-switched network call and, then, routes thevoice-band call as described above for block 412. If the destinationtelephone number is served by an IP gateway, then, the USDS 310 returnsto the OAS 305 the terminating address value called an adjunct logicaladdress (ALA) indicating a partial routing instruction to reach the VoIPnetwork. In block 420, the OAS 305 receives the ALA partial routinginstruction. Then, in block 425, the OAS 305 routes the voice-band calland partial routing information to the HAS 307. In block 430, the HAS307 receives the direct-dialed voice-band call and partial routinginformation. The HAS 307 determines through which circuits to send thedirect-dialed voice-band call by looking up in a routing table, which iskept in a memory in the HAS 307, a routing data block (RDB) isassociated with an adjunct logical address (ALA) from the partialrouting information. Then, in block 435, the HAS 307 routes thedirect-dialed voice-band call to the originating IP gateway 215. Then,in block 440, the HAS 307 performs a call detail recording function andinitiates the VoIP billing record, for example, creates a call detailrecord to track the direct-dialed voice-band call and to be added to atelephone-bill associated with the calling party's telephone number. Aspart of this call detail record, the HAS 307 adds a special moduleincluding the terminating access identification (ID) field, which isbased on information provisioned on the trunk (circuit) groups so thatthe IP network use is explicitly indicated for both billing and trackinguse. The HAS 307 then releases the call detail record to the standardtelephone network billing system. In block 445, the direct-dialedvoice-band call is received by the originating IP gateway 215. In block450, the direct-dialed voice-band call setup is converted to packets. Inblock 455, the partial routing information is used by the originating IPgateway to determine the specific routing information for thedestination IP gateway. In block 460, the packets are routed to thedestination IP gateway over the IP network 110.

In block 465, the packets are received at the destination IP gateway,for example, IP gateway 130, and, then in block 470, the packets arereassembled in their correct order and converted to a reassembledvoice-band call. In block 475, the reassembled voice-band call is routedto the local-access-provider network 340. In block 480, thelocal-access-provider network 340 receives the reassembled voice-bandcall and routes the reassembled voice-band call to the destinationtelephone number 136. In block 485, the reassembled voice-band call isreceived, connected to and conducted by the destination telephonenumber. Then, in block 487, the VoIP call is in progress withtransmissions going back and forth between the calling party's telephonenumber and the destination telephone number. In block 490, the HAS 307receives a notice of call clearing after the VoIP call is completed,which can be signaled when one or both of the calling party and calledparty disconnect from the VoIP call or some portion of the network dropsthe VoIP call. In block 495, the telephone billing system also receivesnotice of the call clearing and closes the call detail record.

For the VoIP service to operate each customer desiring to place VoIPcalls must register/sign-up for the VoIP service and have the systemstore a VoIP service registration record for the calling party prior tothe calling party placing a direct-dialed voice-band call. However, thissign-up process is only done once, just as with other calling plans.Similarly, another preliminary activity involves storing an allowabledestination number list, which identifies numbers accessible using theVoIP service, prior to the calling party placing the direct-dialedtelephone call.

While the above detailed description of the method of operation of theVoIP network system has been described in reference to the embodimentillustrated in FIG. 3, it is not intended to limit the scope of theinvention as other embodiments are contemplated in accordance with thepresent invention.

FIG. 5 depicts a detailed functional flow diagram of the operationdetailed in FIG. 4 for the direct-dialed VoIP apparatus of the entiredirect-dialed VoIP network system of FIG. 3 according to an embodimentto the present invention.

In FIG. 5, in block 505 a direct-dialed voice-band call is received bythe OAS 305 from the calling party's telephone number, where thedirect-dialed voice-band call is associated with a destination phonenumber. In block 510, a test is performed using the USDS 310 todetermine if the calling party's telephone number was registered for theVoIP service. If the calling party's telephone number was not registeredfor the VoIP service, then, in block 512, the voice-band call is routedas a circuit-switched network call and, in block 514, a standardcircuit-switched billing record is initiated. If the calling party'stelephone number was registered for the VoIP service, then, in block515, a test is performed using the USDS 310 to determine if thedestination number of the voice-band call is served by an IP gateway. Ifthe destination number of the voice-band call is not served by an IPgateway, then, in block 512, the voice-band call is routed as acircuit-switched network call and, in block 514, a standardcircuit-switched billing record is initiated. If the calling party'stelephone number was registered for the VoIP service, then, in block 520partial routing information is received by the OAS 305 and, in block525, the direct-dialed voice-band call and partial routing informationare routed to the HAS 307 from the OAS 305. In block 530, thedirect-dialed voice-band call and partial routing information arereceived by the HAS 307. In block 535, the direct-dialed voice-band callis routed to the originating IP gateway 215 by the HAS 307. In block540, the HAS 307 initiates an IP billing record by creating a calldetail record to track the direct-dialed voice-band call and be added toa telephone-bill associated with the calling party's telephone number.In block 545, the HAS 307 receives notice of the destination telephonenumber connecting to the direct-dialed voice-band call. Then, in block550, the VoIP call is in progress with transmissions going back andforth between the calling party's telephone number and the destinationtelephone number. In block 555, the HAS 307 receives a notice of callclearing after the VoIP call is completed, which can be signaled whenone or both of the calling party and called party disconnect from theVoIP call or some portion of the network drops the VoIP call. In block560, the telephone billing system also receives notice of the callclearing and closes the call detail record.

While the above detailed description of the method of operation of theVoIP apparatus has been described in reference to the embodimentillustrated in FIG. 3, it is not intended to limit the scope of theinvention as other embodiments are contemplated in accordance with thepresent invention.

FIG. 6 depicts a detailed block diagram of an offline provisioningsystem that can be configured to operate with the direct-dialed VoIPservice, in accordance with an embodiment to the present invention. InFIG. 6, a network provisioning platform (NPP) 605, which is configuredto receive a customer order for the VoIP service and keep the networkand the billing systems synchronized, is communicatively linked to abilling system 610, which is configured to maintain customer accountinformation and the VoIP calling plan. The NPP 605 is alsocommunicatively linked to a Customer Service Message System (CSMS) 615,which is configured to provision customer calling number and service inthe VoIP network. The CSMS 615 is also communicatively linked to aUniversal Subscriber Data Structure (USDS) database 620 and a 4ESStelecommunications switch 625. The 4ESS 625 is communicatively linked toa data record system 630, which is configured to store call detailrecords for completed VoIP calls with a terminating access ID equal to“IP”.

Specifically, in FIG. 6, the NPP 605 is configured to receive a VoIPservice registration/order for the calling party's telephone number,generate a billing system order and a network service order for thecalling party's telephone number VoIP service and, then, forward theorders to the billing system 610 and the CSMS 615, respectively. The NPP605 is also configured to manage the interaction between a calling partytelephone number and a billed telephone number, to provide updatedrecords to the billing system 610 and the CSMS 615 to compensate fornumbering plan changes. The NPP 605 is further configured to synchronizeany other changes due to calling party activations, disconnectionsand/or changes in the billing system 610 and the CSMS 615.

The billing system 610 is further configured to maintain calling partytelephone number account information, to maintain a rating table withapplicable PSTN and VoIP rates, and to create bills using the ratingtable, use records, terminating access ID and calling plan uniformservice order code (USOC).

The CSMS 615 is further configured to synchronize between the 4ESStelecommunications switch 625 and the network database in the USDS 620,which stores the calling party telephone numbers that are registered forthe VoIP service, USOC information and destination number information.

While the above detailed description of the offline provisioning systemhas been described in reference to the embodiment illustrated in FIG. 6,it is not intended to limit the scope of the invention as otherembodiments are contemplated in accordance with the present invention.

FIG. 7 depicts a detailed functional flow diagram of the operation of ascalable offline provisioning system in FIG. 6 according to anembodiment to the present invention. The system is scalable because itcan be configured to work with both small (for example, a corporatetelephone network on a LAN) and large systems (for example, the nationaland international system illustrated in FIG. 3). In FIG. 7, in block705, the NPP 605 receives a VoIP service registration/order for acalling party telephone number and, then in block 710, the NPP 605generates a billing system order and a network service order for thecalling party telephone number's VoIP service. In block 715, the NPP 605forwards the billing system order to the billing system 610 to be storedand the network system order to the CSMS 615 to be stored. In block 720,the NPP 605 manages the billing interaction for a billed account betweenat least one calling party telephone number and a billed telephonenumber when a call is placed from the calling party telephone number.That is, the NPP 605 ensures that the call detail record generated foreach call is billed to the correct telephone number after the call isterminated. In general, the calling party telephone number and thebilled telephone number are the same number, however, these numbers canalso be different numbers. In block 725, the NPP 605 provides updatedrecords to the billing system 610 and the CSMS 615 to reflect numberingplan changes. In block 730, the NPP 605 synchronizes the changes betweenthe billing system 610 and the CSMS 615 due to, for example, callingparty activations, disconnections and other changes. In block 735, thebilling system 610 processes the call detail records from completedcalls, where the call detail records include information on theterminating access ID, VoIP rating and bill notification.

The CSMS 615 is further configured to synchronize between thetelecommunication switch 625 and the network database in the USDS 620,which stores the calling party telephone numbers that are registered forthe VoIP service, network service information (for example, USOCinformation) and destination number information.

While the above detailed description of the method of operation of theoffline provisioning system has been described in reference to theembodiment illustrated in FIG. 6, it is not intended to limit the scopeof the invention as other embodiments are contemplated in accordancewith the present invention. For example, embodiments of the present canbe implemented to handle intra-state, inter-state and internationalvoice-band calls (for example, regular telephone calls, facsimiletransmissions and modem initiated calls) using standard circuit-switchedtelephone lines, cable, twisted pair, digital subscriber line (DSL) andwireless. Similarly, in an embodiment of the present invention, thecustomer can register for the VoIP service by selecting both a provider,that is a Primary Interexchange Carrier (PIC) and a calling plan.Alternatively, in another embodiment of the present invention, thecustomer can register for the VoIP service by only selecting a PIC, forexample, if the PIC does not need to have a calling plan associated withthe PIC in order to provide the VoIP service. The select the PIC onlyembodiment could be used, for example, for a provider that only providesthe VoIP service.

In an embodiment of the present invention, a method for routingdirect-dialed voice-band calls over an IP network includes receiving adirect-dialed voice-band call from a calling party telephone number, thedirect-dialed voice-band call being associated with a destinationnumber. The method further includes automatically routing thedirect-dialed voice-band call to the destination number as a VoIPtelephone call if the calling party telephone number is registered for aVoIP service and if the destination number of the direct-dialedtelephone call is accessible by the VoIP service.

In an embodiment of the present invention, a method for automaticallyprovisioning and maintaining a network system for routing direct-dialedvoice-band calls from a calling party telephone number over an IPnetwork includes receiving a VoIP service registration for the callingparty telephone number, generating at least one order record for thecalling party telephone number's VoIP service and storing the at leastone order record for the calling party telephone number's VoIP service.The method further includes managing the billing interaction for abilled account between at least one calling party telephone number and abilled telephone number; synchronizing changes made to the stored atleast one order record for the calling party telephone number's VoIPservice, between the network system and a billing system, due to callingparty activations, disconnections and changes; and processing at leastone call detail record including at least a terminating accessidentification (ID).

In an embodiment of the present invention, an apparatus includes a firstvoice-band switch and a database coupled to the first voice-band switch.In the apparatus, the first voice-band switch is configured to receive adirect-dialed voice-band call from a calling party's telephone numberand to automatically designate the direct-dialed voice-band call as aVoIP call.

In an embodiment of the present invention, an apparatus includes an ESSOAS, with the OAS being configured to receive a direct-dialed voice-bandcall from a calling party's telephone number, the direct-dialedvoice-band call being associated with a destination telephone number.The OAS is further configured to determine whether to route thedirect-dialed voice-band call over an IP network or a circuit-switchednetwork and, if it is determined to route the direct-dialed voice-bandcall over the IP network, the OAS is configured to transmit thedirect-dialed voice-band call to the IP network, or, if it is determinedto continue to route the direct-dialed voice-band call over thecircuit-switched network, the OAS is configured to transmit thedirect-dialed voice-band call to the circuit-switched network. Theapparatus further includes an USDS coupled to the ESS OAS, with the USDSbeing configured to store information on a plurality of calling party'stelephone numbers registered for the VoIP service, to store informationon which destination telephone numbers are accessible using the VoIPservice, to receive the calling party's telephone number and adestination telephone number of the direct-dialed voice-band call fromthe OAS, to determine if the calling party's telephone number isregistered for the VoIP service, and, if the calling party's telephonenumber is registered for the VoIP service, to determine if thedestination telephone number is accessible using the VoIP service, andto return a partial routing instruction and service information to theOAS.

In an embodiment of the present invention, a system for automaticallyprovisioning and maintaining a network system for routing direct-dialedvoice-band calls from a calling party telephone number over an IPnetwork includes a network provisioning component. The networkprovisioning component is configured to receive a VoIP serviceregistration for the calling party telephone number, to generate atleast one order record for the calling party telephone number's VoIPservice, to store the at least one order record for the calling partytelephone number's VoIP service, to manage the billing interaction for abilled account between at least one calling party telephone number and abilled telephone number, and to update the at least one order record tocompensate for numbering plan changes. The system further includes abilling system component coupled to the network provisioning component,and the billing system component is configured to maintain at least onecalling party's account information, to maintain the VoIP service, tocreate bills based on usage, terminating access ID and calling planUSOC. The system further includes a CSMS component, which is coupled tothe network provisioning component. The CSMS component is configured tosynchronize changes made to the stored at least one order record for thecalling party telephone number's VoIP service, between at least onetelecommunications switch and a database, which stores calling partytelephone numbers that are registered for the VoIP service, USOCinformation and destination number information. The network provisioningcomponent is further configured to synchronize changes made to thestored at least one order record for the calling party telephonenumber's VoIP service, between the network system and a billing system,due to calling party activations, disconnections and changes.

In the foregoing detailed description and figures, several embodimentsof the present invention are specifically illustrated and described.Accordingly, it will be appreciated that modifications and variations ofthe present invention are covered by the above teachings and within thepurview of the appended claims without departing from the spirit andintended scope of the invention.

What is claimed is:
 1. A method for routing direct-dialed voice-bandcalls over an internet protocol network, comprising: receiving adirect-dialed voice-band call from a calling party telephone numberassociated with a telephone attached to a circuit-switched publicswitched telephone network, the direct-dialed voice-band call beingassociated with a destination number, wherein the calling partytelephone number is associated with a telephone bill; determining if thedestination number is served by an internet protocol gateway bysearching an internet protocol command center database in communicationwith an internet protocol gateway serving the calling party telephonenumber; routing the direct-dialed voice-band call to the destinationnumber that is served by the internet protocol gateway as avoice-over-Internet protocol telephone call, via a direct link from alocal access provider network to an internet protocol gateway thatbypasses the public switched telephone network located between the localaccess provider network and the internet protocol gateway, if thecalling party telephone number is registered for a voice-over-internetprotocol service and if the destination number of the direct-dialedtelephone call is accessible by the voice-over-internet protocolservice; generating a voice-over-internet protocol billing record for avoice-over-internet protocol portion of the direct-dialed voice-bandcall via a switch in the public switched telephone network; and addingthe voice-over-internet protocol billing record to the telephone bill.2. The method of claim 1 further comprising: receiving a registrationfor the calling party telephone number for the voice-over-internetprotocol service prior to the calling party placing the direct-dialedtelephone call.
 3. The method of claim 2 further comprising: storing avoice-over-internet protocol service registration record for the callingparty telephone number.
 4. The method of claim 2 further comprising:storing an allowable destination list, which identifies the destinationnumbers accessible using the voice-over-internet protocol service, priorto the calling party placing the direct-dialed telephone call.
 5. Themethod of claim 1, wherein the routing the direct-dialed voice-band callto the destination number as a voice-over-Internet protocol telephonecall if the calling party telephone number is registered for avoice-over-internet protocol service and the destination number of thedirect-dialed telephone call is accessible by the voice-over-internetprotocol service comprises: determining if a voice-over-internetprotocol service registration record for the calling party telephonenumber exists; and if the voice-over-internet protocol serviceregistration record for the calling party telephone number exists,determining if the destination number of the direct-dialed voice-bandcall is accessible by the voice-over-internet protocol service.
 6. Themethod of claim 5 further comprising: if the calling party telephonenumber is registered for the voice-over-internet protocol service and ifthe destination number of the direct-dialed voice-band call isaccessible by the voice-over-internet protocol service, receiving anindication to route the direct-dialed voice-band call over the internetprotocol network, otherwise, receiving an indication to route thedirect-dialed voice-band call over the circuit-switched public switchedtelephone network.
 7. The method of claim 6 further comprising:initiating a billing record for the direct-dialed voice-band call if theindication is to route the direct-dialed voice-band call over theinternet protocol network, wherein the billing record is associated withthe calling party telephone number.
 8. The method of claim 7 furthercomprising: routing the direct-dialed voice-band call to the internetprotocol network.
 9. The method of claim 8 further comprising: receivingnotice of the direct-dialed voice-band call clearing.
 10. The method ofclaim 9 further comprising: closing the billing record for thedirect-dialed voice-band call.
 11. A switch for routing direct-dialedvoice-band calls over an Internet protocol network, the switchconfigured to: receive a direct-dialed voice-band call from a callingparty telephone number associated with a telephone attached to acircuit-switched public switched telephone network, the direct-dialedvoice-band call being associated with a destination number, wherein thecalling party telephone number is associated with a telephone bill;determine if the destination number is served by an internet protocolgateway by searching an internet protocol command center database incommunication with an internet protocol gateway serving the callingparty telephone number; route the direct-dialed voice-band call to berouted to the destination number that is served by the internet protocolgateway as a voice-over-Internet protocol telephone call, via a directlink from a local access provider network to an internet protocolgateway that bypasses the public switched telephone network locatedbetween the local access provider network and the internet protocolgateway, if the calling party telephone number is registered for avoice-over-internet protocol service and if the destination number ofthe direct-dialed telephone call is accessible by thevoice-over-internet protocol service; generate a voice-over-internetprotocol billing record for a voice-over-internet protocol portion ofthe direct-dialed voice-band call in the public switched telephonenetwork; and add the voice-over-internet protocol billing record to thetelephone bill.
 12. The switch of claim 11, wherein the switch isfurther configured to: receive a registration for the calling partytelephone number for the voice-over-internet protocol service prior tothe calling party placing the direct-dialed telephone call.
 13. Theswitch of claim 12, wherein the switch is further configured to: store avoice-over-internet protocol service registration record for the callingparty telephone number.
 14. The switch of claim 12, wherein the switchis further configured to: store an allowable destination list, whichidentifies the destination numbers accessible using thevoice-over-internet protocol service, prior to the calling party placingthe direct-dialed telephone call.